providing a high-quality service for transmission and playing real-time voice conversations (voice streaming) over wireless ad-hoc networks is no mean feat. buffering together with adjusting the playout time of the packets is a receiver-side solution to overcome this challenge. in this paper, a new adaptive playout adjustment algorithm is proposed to stream the voice conversations over wireless...

Much effort has been involved in packet-level error control and delay jitter concealment for Internet audio applications. A packet-based time-scale modification scheme for speech signals is developed to provide adaptive delay concealment at the receiver for an Internet voice session in this work. The adaptive playout algorithm strives to minimize packet droppings for late-arrival packets and pr...

A novel playout algorithm for VoIP applications is presented. The playout times of voice packets are calculated using adaptive estimation of network delays. In contrast to previous solutions, the weighting factor that controls the estimation process is dynamically adjusted according to the observed delay variations. This results in higher-quality estimates of network delays. With this algorithm...

Voice over IP (VoIP) applications are now very popular and widely used on the Internet. Such applications use receiver playout buffers to smooth delay variations so as to reconstruct the periodic form of the transmitted packets. Packets arriving after their scheduled playout time are considered late and are not played out. Playout delay control algorithms often operate by updating the playout d...

Transmitting real-time audio/video over the Internet is very difficult due to packet loss and jitter. These parameters vary depending on the locations of the senders and receivers, with typical packet loss rates of 0−20% and one-way delays of 5-500 ms. Delay variations occur within and across audio and video streams, complicating the synchronization process. One possibility for reducing jitter ...

Providing a high-quality service for transmission and playing real-time voice conversations (voice streaming) over wireless ad-hoc networks is no mean feat. Buffering together with adjusting the playout time of the packets is a receiver-side solution to overcome this challenge. In this paper, a new adaptive playout adjustment algorithm is proposed to stream the voice conversations over wireless...

Network variations in video streaming require to prebuffer sufficient data at the client prior to playout. This receiver buffer prevents the display process from starvation in case of network congestion. Prebuffering, however, is also responsible for the major part of the delay between requesting a media stream and playing it at the receiver. In this paper we show how adaptive media playout can...

In Voice over IP, the quality of interactive conversation is important to users. Major factors affecting perceived quality are delay, delay jitter, and missing packets. For conversational VoIP, a conversational delay also plays an important role for perceived quality. Large conversational delay can result in double talk, echo or even the termination of the conversation. In practice, a playout b...

Delivering PSTN-like quality over current besteffort Internet infrastructure presents many technical challenges. Much research in recent years has focused on receiver-based approaches which adapt to varying network conditions in order to optimize playout quality. In this paper, we propose and evaluate a receiver-based approach that implements a hybrid adaptivefixed playout regime by integrating...

Recent interest in supporting packet-audio applications over wide area networks has been fueled by the availability of low-cost, toll-quality workstation audio and the demonstration that limited amounts of interactive audio can be supported by today's Internet. In such applications, received audio packets are bu ered, and their playout delayed at the destination host in order to compensate for ...